EP1665881B1 - Verfahren zur steuerung der richtcharakteristik eines hörgeräts und signalverarbeitungsvorrichtung für ein hörgerät mit steuerbarer richtcharakteristik - Google Patents

Verfahren zur steuerung der richtcharakteristik eines hörgeräts und signalverarbeitungsvorrichtung für ein hörgerät mit steuerbarer richtcharakteristik Download PDF

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EP1665881B1
EP1665881B1 EP03769294A EP03769294A EP1665881B1 EP 1665881 B1 EP1665881 B1 EP 1665881B1 EP 03769294 A EP03769294 A EP 03769294A EP 03769294 A EP03769294 A EP 03769294A EP 1665881 B1 EP1665881 B1 EP 1665881B1
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Prior art keywords
signal
signals
directional
output
processing apparatus
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French (fr)
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EP1665881A1 (de
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Lars Baekgaard Jensen
Kristian Tjalfe Klinkby
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Widex AS
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Widex AS
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/40—Arrangements for obtaining a desired directivity characteristic
    • H04R25/407—Circuits for combining signals of a plurality of transducers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/40—Arrangements for obtaining a desired directivity characteristic
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
    • H04R2225/43—Signal processing in hearing aids to enhance the speech intelligibility
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/40—Arrangements for obtaining a desired directivity characteristic
    • H04R25/405—Arrangements for obtaining a desired directivity characteristic by combining a plurality of transducers

Definitions

  • the present invention generally relates to a hearing aid with a controllable directional characteristic and more particular to a method for controlling the directionality of the sound receiving characteristic for minimizing noise and to a signal processing apparatus for carrying out the method.
  • acoustical signal-to-noise ratio can be significantly improved by e.g. using of dedicated directional microphones or equivalently by a pair of omni-directional microphones followed by a delay and subtracting procedure to employ a directional sound receiving characteristic.
  • Hearing aids with more than two microphones have also been developed in the pursuit of highly selective directionality.
  • Hearing aids having a directional sound receiving characteristic are useful to improve speech perception in noisy environments, where human speech may be received simultaneously from different directions, as is the case e.g. in the noise environment frequently referred to as the so-called cocktail party noise.
  • the speech perception in a hearing aid is improved by reduced perception of sound coming from the back and the sides of the user while maintaining the level of sound coming from the area in front of the user.
  • the hearing aid user will normally prefer an omni-directional or spherical sound receiving characteristic offering the same perception of sound irrespectively of the direction, from which it arrives.
  • hearing aids with adaptive directional functionality have been introduced with the aim to place significant damping in the direction of the dominant noise source.
  • WO 01/01731-A1 such a method for controlling the directionality of a sound receiving characteristic of the hearing aid is disclosed.
  • the hearing aid comprises spaced apart microphones, wherein the sound receiving characteristic may change between an omni-directional characteristic and a directional characteristic.
  • an adjustable time or phase delay may be imposed.
  • the directional characteristic may be created by adjusting the delay of a delay device to be the same as the acoustical delay between the back microphone and the front microphone. With this delay, signals that are first received at the back microphone and are later received at the front microphone, are suppressed in an adding circuit, where the delayed signal of the back microphone is subtracted from the output signal of the front microphone.
  • the hearing aid may exercise a smooth change-over between an omni-directional characteristic and a directional characteristic, substantially without changing the phase relationship or time delay and the amplitude characteristic of the signal.
  • Both the fixed and the adaptive directional functions suffer from a reduced signal-to-microphone noise ratio because of lack of low frequency sensitivity for acoustical signals, since one consequence of adding a signal (from the front microphone) with its delayed and inverted replicate (from the back microphone) to achieve a directional advantage is that the sensitivity of the microphone at low frequencies is reduced also for sounds presented directly in front of the listener. For a given delay and distance between microphones, the low frequency sensitivity rolls off at a rate of 6 dB per octave.
  • Present adaptive systems like the directional controller disclosed in WO 02/085066-A1 adjusts the directional characteristic by minimizing the output signal of the system. Since signals coming from the frontal direction are not affected by changing the directional characteristic of the system, a minimization of the output signal results in damping of undesired noise and an improvement of the signal-to-noise-ratio. However, such a signal-to-noise-ratio optimization applies only if the desired signals are coming from the frontal direction and noise signals are coming from another direction.
  • the speech may very well be a desired signal.
  • the above described adaptive systems will try to damp this speech signal in order to minimize the output signal, and thereby increase the microphone noise.
  • the adaptive system will try to damp the microphone noise. This results in dynamically undesired damping of the actual desired signal and a significant modulation of the microphone noise reducing speech perception and sound quality.
  • a supposed solution to the problem seems to be to modify the microphone signals as input signals for the adaptive function or to modify the output signal as the control signal in the adaptive function.
  • Such modifications have the following drawbacks.
  • One problem is that the possible modifications of signals in the signal path, e.g. filtering away undesired frequency areas, are very limited, because the following adaptation algorithm needs the gain and the delay information of the input signals to be able to adapt correctly. For this reason, a signal modification that e.g. just leaves the envelope of the two microphone signals is not possible.
  • Another problem is that adaptive systems generally should adapt the output signal relatively soon after the input signals have changed. If not, the system would adjust the characteristic only after a certain delay in which the system is not correctly adapted.
  • adaptive systems generally should receive the response to a parameter change relatively soon after the parameter has changed. If not, the system would change the parameter further in a certain direction, before getting the response that the parameter change in this direction was in fact erroneously. As a result, such an adaptive system with a delayed response will not reach its optimum very precisely, if at all, and may become unstable.
  • Algorithms for separating signals with different characteristics e.g. separating noise from desired speech signals
  • the hearing aid has an adaptive directional functionality and comprises a speech detector. On the detection of speech simultaneously with noise, the hearing aid switches from omni-directional mode into directional mode with reception in the direction of vision or fixes in directional mode the sensitivity at a direction of vision.
  • the present invention overcomes the foregoing and other problems by providing an adaptive directional function which minimizes only undesired signals, e.g. undesired noise.
  • Signals that comprise wanted signals like speech signals are herein after referred to as desired signals.
  • a signal processing apparatus a method of controlling the directional characteristic of a hearing aid, a hearing aid, a computer program product, and a software tool as defined by the independent claims.
  • Methods, apparatuses, systems and articles of manufacture consistent with the present invention use a detecting mechanism to detect that only undesired signals are submitted as input signals to the adaptive directional function, and the adaptive directional function then adjusts the directionality of a sound receiving characteristic in order to minimize the output signal of the adaptive directional function.
  • the detection mechanism detects that the input signals to the adaptive directional function also comprise desired signals, adjustment of the directionality of the sound receiving characteristic of the adaptive directional function is stalled for a certain amount of time.
  • the adaptive directional function receives an additional control signal from a signal analyzer that effectively provides a desired signal detector (DSD).
  • the DSD generates this additional control signal for the adaptive directional function, which allows one or more of the original control parameters to be updated only if the DSD concludes that the input signals to the adaptive directional function are undesired signals. If the DSD concludes that the input signals are desired signals or a mixture of desired and undesired signals, the control parameters of the adaptive directional function will not be updated, and the adaptation is stalled.
  • the adaptive directional function works on unmodified input signals and further requires no modification of the fed back output signal.
  • the additional control signal submitted by the DSD indicates to stall or to update the adaptation in the adaptive directional function.
  • the additional control signal is generated in the DSD outside the main signal path between input and output signals, so that the generation of the additional control signal may be done in different, the input signals distorting and also very complex ways without affecting the quality of the output signal.
  • the DSD may use statistical analysis of the input time signal, distinguish between high and low frequency signals, detect whether the input signal level is above or below a certain fixed limit, detect whether the incoming signals are sufficiently correlated, or distinguish between desired and undesired signals by applying any other suitable decision rule.
  • the adaptive directional function is to be understood as a directional controller receiving at least first and second microphone signals supplied by a first (front) microphone and a second (back) microphone as input signals and which outputs an output signal, wherein the output signal is generated by combining the first and second microphone signal according to the present directional characteristic adjusted by the directional controller.
  • a method for controlling the directional characteristic of a hearing aid has spaced apart front and back microphones, a directional controller receiving first and second microphone signals supplied by the front and back microphones, respectively, and outputting an output signal, wherein the output signal is generated by combining the first and second microphone signals according to the directional characteristic.
  • a signal processing apparatus for a hearing aid with a controllable directional characteristic comprising a directional controller for receiving first and second microphone signals and outputting an output signal and a signal analyzer for detecting whether at least one of the first and second microphone signals are undesired signals.
  • the directional controller receives first and second microphone signals submitted by e.g. a front and a back microphone, respectively, and outputs an output signal.
  • the signal analyzer that effectively provides a desired signal detector determines whether the first and/or second microphone signals are undesired signals, and the directional controller minimizes the output signal by adjusting the directional characteristic only if the desired signal detector has detected undesired signals.
  • the signal processing apparatus according to the present invention and a conventional system behave nearly similar, but when the speaker moves e.g. to one side of the user, the apparatus according to the invention will avoid attempts at trying to adjust the directional characteristic in order to minimize the output signal with the risk of suppressing the speaker.
  • the signal processing apparatus with the desired signal detector stays basically in omni-directional characteristic also when the speaker moves to one side of the user, because the DSD forces the directional controller not to optimize its directional characteristic while the speaker sentences and only allows the directional controller to adapt the directional characteristic during the pauses when the speaker does not sentence.
  • the directional controller only tries to minimize the microphone noise which is dominant during the pauses, and which is best done by staying in omni-directional characteristic.
  • the microphone noise stays low and is not fluctuating and a desired signal coming from one side of the user is not damped so that speech perception and sound quality is improved.
  • a hearing aid with a controllable directional characteristic comprises an adaptive directional function of which the adaptation is stalled for a certain amount of time if desired signals have been detected as input signals submitted by spaced apart first and second sound receiving means.
  • the processed input signals are output as a combined output signal by the adaptive directional function.
  • the hearing aid further comprises an output transducer for emission of sound signals in response to the output signal.
  • the invention also provides a software tool for implementing a directional controller on a signal processing apparatus and a computer program product comprising computer program code which, when executed on a computer or a signal processing system, enables the computer or signal processing system to carry out a method according to the present invention.
  • the methods, systems and articles of manufacture consistent with the present invention are preferably used in all kinds of hearing aids having a directional characteristic (e.g., behind the ear (BTE), in-the-ear (ITE), in-the-channel (ITC)) for all degrees of hearing loss to improve the ability of a user to understand desired signals like voice or speech signals or sound signals emitted by a radio or TV or other.
  • desired signals may come from any direction of the user.
  • Fig. 1 depicts a block diagram of a signal processing apparatus 100 suitable for a hearing aid with a controllable directional characteristic and for practicing methods and implementing a system consistent with an embodiment of the present invention.
  • the signal processing apparatus 100 comprises a directional controller 10 which receives first and second microphone signals 20, 30 and outputs output signal 40.
  • First and second microphone signals may submitted by a first front microphone F mic and a second back microphone B mic directly or via preprocessing function, e.g. a filter function.
  • the output signal 40 may be used as an input signal for a signal processor of the hearing aid for further processing and amplifying the output signal and submitting signals output from said signal processor to an output transducer, e.g. a loudspeaker, for emission of sound signals (not shown in Fig. 1 ).
  • an output transducer e.g. a loudspeaker
  • the directional controller 10 is capable of applying an adaptive directional function 50 based on a processing of the first and second microphone signals 20, 30. As a result of the adaptive directional function 50, the combined output signal 40 is provided.
  • the directional characteristic of the adaptive directional function 50 is adjusted by first and second control parameters 60, 80.
  • First control parameter 60 is the fed back output signal 40.
  • the signal processing apparatus 100 further comprises a signal analyzer, also referred to as a desired signal detector (DSD) 70, receiving first and second microphone signals 20, 30 and outputting second control parameter 80.
  • DSD desired signal detector
  • the desired signal detector 70 may receive just one of the first or second microphone signals as input signal.
  • sounds from the environment of the hearing aid are picked up by both the first front microphone F mic and the second back microphone B mic (not shown).
  • the electrical signals generated by the two microphones may then be preprocessed by a sample unit on a sampling rate of e.g. 32 kHz and further analogue-digital converted by e.g. a 24 bit analogue-to-digital-converter.
  • the resulting digital signals corresponding to the sounds picked up by the microphones are then submitted as first and second microphone signals 20, 30.
  • Fig. 5 shows a method according to the present invention.
  • the directional controller 10 processes the first and second microphone signals 20, 30 according to the adjusted directional characteristic of the adaptive directional function 50 and combines these processed signals to the output signal 40.
  • the adaptive directional function is adjusted by internal delay and attenuation parameters (internal parameters) to delay and attenuate the first and second microphone signals (not shown in Fig. 1 ).
  • the adaptive directional function 50 adjusts the internal parameter such that the fed back output signal 60 is minimized.
  • the adaptive control of the internal parameters in the adaptive directional function by minimizing the output signal is carried out by measurements known in the art, e.g. by applying a so-called LMS-algorithm in the adaptive directional function. Examples of such adaptive control with an LMS-algorithm can be found in e.g. US-A-5,259,033 or US-A-5,402,496 , however, the adaptive control systems provided in these references do not control a directional controller.
  • the desired signal detector 70 detects in operation 510 whether the first and second microphone signals as input signals are undesired signals.
  • an undesired signal comprises only undesired noise and no desired signals like speech signals. If the DSD 70 in operation 510 detects desired signals, then adaptation of the internal parameter is blocked or frozen so that the adaptive directional function does not adjust the internal parameters by adapting them according to the current fed back output signal. Hence, the minimization of the output signal in operation 520 is stalled for a certain amount of time since the output signal is generated in operation 540 without adaptation of the internal parameters.
  • the stall time depends on the input signals and the actual implementation of the DSD. For example, the stall time may have a value in the range of 3 to 30 ms .
  • the adaptation continues and the internal parameters are updated to adjust the directional characteristic in operation 530 in order to minimize the output signal (operation 520).
  • the directional characteristic is only adapted if the DSD detects undesired signals as input signals.
  • Fig. 2 shows a block diagram of a hearing aid 220 according to an embodiment of the present invention.
  • the signal path of the hearing aid 220 comprises first and second input transducers or microphones Fmic and Bmic transforming acoustic input signals into first and second electrical microphone signals 20, 30, a signal processing apparatus 200 with a controllable directional characteristic generating an electrical output signal 40 and an output transducer 210, e.g. a loudspeaker or receiver, for transforming the electrical output signal into an acoustic output signal.
  • the signal processing apparatus 200 comprises a directional controller 10 with first and second microphone signals 20, 30 as input signals and output signal 40.
  • the signal processing apparatus 200 further comprises desired signal detector 70 and parameter controller 90.
  • Parameter controller 90 adjusts internal parameter(s) 95 of adaptive directional function 50 in order to minimize the fed back output signal 60 which is input to parameter controller 90.
  • parameter controller 90 receives second control signal 80 supplied by desired signal detector 70.
  • the desired signal detector 70 receives first and second microphone signals 20, 30 as input signals and further comprises a detector 71 and a update/stall-circuit 72.
  • Detector 71 detects whether first and second microphone signals are undesired signals or not. If the detector 71 detects that the input signals are undesired signals, the update/stall-circuit 72 provides a second control signal 80 which enables the parameter controller 90 to adjust the internal parameter(s) 95 in order to minimize the output signal 40.
  • the update/stall-circuit 72 provides a second control signal 80 that indicates the parameter controller 90 to disable or stall the adaptation process and not to minimize the output signal further until the detector 71 detects again undesired signals.
  • Fig. 3 shows a directional controller according to WO 01/01731-A1 which may be implemented as directional controller 10 in a signal processing apparatus 100, 200 according to the present invention.
  • controllable attenuation and phase delay operations are applied to signals X front , X back from front and back microphones F mic and B mic corresponding to first and second microphone signals 20, 30.
  • the resulting signals are then combined to an output signal corresponding to output signal 40.
  • the directional controller carries out an adaptive directional function and comprises a first adding circuit 12 connected with the front and back microphones F mic and B mic and a first subtraction circuit 13 having a positive input connected with the front microphone F mic and a negative input connected with back microphone B mic .
  • First and second phase delay devices 14 and 15 are connected with the first subtraction and adding circuit 13 and 12, respectively.
  • the second adding circuit 16 is connected with the first subtraction circuit 13 and the first phase delay device 14 and a second subtracting circuit 17 has its positive input connected with the first adding circuit 12 and its negative input connected with second phase delay device 15.
  • a first controllable attenuator 18 acts on the signal from the second adding circuit 16 for attenuation of this signal by a factor (1 - omni) / 2 and a second controllable attenuator 19 acts on the signal from the second subtraction circuit 17 for attenuation of this signal by a factor (1 + omni) / 2, whereas a third adding circuit 21 is connected with the first and second attenuators 18 and 19 for addition of the signals therefrom to provide the overall combining signal to be supplied to the signal processor.
  • the properties of this directional controller are such that it may advantageously be utilized in connection systems and methods according to the present invention.
  • omni is an adjustable internal parameter 95, controlling attenuators 18 and 19 and having in the implementation of WO 01/01731-A1 a value in the range from 0 to 1.
  • acoustical delay between the back microphone B mic and the front microphone F mic is designated A
  • X back X front * e - j ⁇ A
  • the internal parameter omni may assume values outside the range of 0 to 1.
  • Fig. 4 shows an embodiment of a desired signal detector 70 according to an embodiment of invention.
  • the desired signal detector 70 may be used in a signal processing apparatus 100, 200 as described with reference to Fig. 1 and 2 .
  • the circuit structure of the desired signal detector comprises an adding circuit 73 for adding first and second microphone signals 20, 30, which are connected to the adding circuit 73.
  • the output of the adding circuit is connected to a signal envelope circuit 74 which produces the signal envelope of the added input signals.
  • the signal envelope as output of the signal envelope circuit 74 is submitted to both a comparator 77 and a percentile estimator circuit 76.
  • the percentile estimator circuit 76 generates a percentile estimator result, e.g. a 10 % or 11 % estimator result of the signal envelope.
  • percentile estimators It is well known to a skilled person how to provide such an percentile estimator result with an percentile estimator known in the art. Examples of such percentile estimators are known from e.g. US-A-4,204,260 , WO 95/15668 , or WO 98/27787 , however, these percentile estimators are not part of a desired signal detector.
  • the percentile estimator result output by the percentile estimator 76 may be any percentile estimator result in the range 0 - 100 %.
  • 0 % percentile estimator result means that all signals input to the percentile estimator are detected to be above the percentile estimator result and will thus be considered as speech. This means the DSD detects desired signals all the time and the DSD causes the adaptation to not run at all. The other extreme, if the percentile estimator result is 100 %, all signals input to the percentile estimator are detected to be below the percentile estimator result. This means the DSD considers the input signals as undesired signals so that the DSD will not stall the adaptation at all, and the directional adaptation will run as if the DSD was not present.
  • the percentile estimator result is not necessarily limited, for most applications a number between 5 - 90 % is selected.
  • the percentage used for the DSD is not limited to a specific number, but there are however some practical limitation depending on the surrounding noise situation.
  • the percentile estimator result should generally present a good border level between noise and speech (undesired and desired signals), so that levels below the percentile estimator result can be considered as essentially undesired signals and levels above can be considered as comprising desired signals. If the percentage is set too high, some part of the speech signal is below the percentile estimator result and will incorrectly be considered as noise. The adaptation will therefore not be stalled in every necessary occasion, and hence the directional adaptation will to some degree react on the speech as well as the noise.
  • a low percentage percentile estimator e.g. in the range 5 - 20 %, will find the noise floor quite well, but the final choice will always be a matter of trade-offs, because different sound environments may yield different optimal values.
  • a DSD 70 having a percentile estimator with a percentile estimator result between 10 - 20 % good results could be achieved by processing first and second microphone signals 20, 30 supplying speech signals of a single speaker in a quiet room.
  • the percentile estimator result as output of the percentile estimator 76 is supplied as second input signal to comparator 77.
  • Comparator 77 compares the two input signals, the signal envelope submitted by signal envelope circuit 74 and the percentile estimator result. The result of the comparison is submitted to an update/stall-circuit 72 which produces the second control parameter 80.
  • DSD desired signal detector
  • the signal envelope is generated from said input signals.
  • the input signals may be the added first and second microphone signals 20, 30 according to Fig. 4 .
  • the desired signal detector does not comprise adding circuit 73 and the input signal to the signal envelope circuit 74 is either the first or the second microphone signal.
  • the adding circuit may be left out according to the presumption that at least one of the first front microphone F mic or the second back microphone B mic is a microphone with an omni-directional characteristic so that this microphone submits a microphone signal corresponding to the sound signals reaching that microphone from any direction.
  • the signal envelope of the sound signals surrounding the user may be generated from only one microphone signal in order to keep the overall circuitry more simple.
  • a percentile estimator result e.g. a 10 % percentile estimator result
  • the signal levels of both signals, the percentile estimator result and the signal envelope, are then compared in operation 630.
  • comparator 77 detects when the instantaneous signal of the signal envelope goes above the percentile estimator result and also when the instantaneous signal of the signal envelope goes below the percentile estimator result (operation 640).
  • the desired signal detector concludes "desired signals" in the input signals and the control parameter adjustment is stalled in operation 650.
  • update/stall-circuit 72 submits second control parameter 80 indicating to the parameter controller 90, or directly to the directional controller 10, to disable adaptation of the directional characteristic by the adaptive directional function 50.
  • the desired signal detector concludes "undesired signal" as input signal and allows to update the control parameter adjustment in operation 660 by setting an enable-signal as second control parameter 80 to adjust the directional characteristic by adjusting the internal control parameter for the adaptive directional function in operation 670.
  • the desired signal detector comprises filter circuitry which is capable to distinguish between high and low frequencies in the input signals 20, 30. If the detector 71 then detects that the input signals comprise signals in a certain frequency range, e.g. corresponding to voice signals, the digital signal detector concludes “desired signal” and proceeds with operation 650. Otherwise, if the detector 71 detects that the input signals are outside a certain frequency range, the digital signal detector concludes "undesired signal” and proceeds with operation 660.
  • detector 71 detects the level of the input signal and in an operation similar to operation 640 it is decided whether the level of the input signal is above or below a certain preset level, and if the input level is below that preset level, the DSD concludes "undesired signal” and proceeds with operation 660 and vice versa.
  • Fig. 9 shows an embodiment of a desired signal detector 170 according to an embodiment of the invention in which such a level detection is implemented to distinguish between desired and undesired signals.
  • the circuit structure of the DSD 170 is similar to the one of the DSD 70 described with reference to Fig. 4 .
  • the DSD 170 comprises a level generator 110 which replaces the percentile estimator 76 of DSD 70.
  • the level generator 110 does not necessarily need any input, e.g. the envelope signal output by signal envelope circuit 74, but provides a fixed signal level to the comparator 77 which compares the two input signals, the signal envelope submitted by signal envelope 74 and the level submitted by the level generator 110. The result of the comparison is submitted to the update/stall-circuit 72 which again produces the second control parameter 80.
  • the function of the DSD 170 is also similar to the function of the DSD 70, except for the fact, that the level generator 110 outputs a fixed signal level which does not depend on the signal envelope of the input signals 20, 30. Therefore, in operation 640, it is decided whether the level of the instantaneous signal envelope is above or below the signal level of the level generator.
  • the value of the signal level generated by the level generator 110 and serving as a threshold and the update and stall criteria may be adjusted accordingly.
  • the designer might want to use this capability to disable adaptation below a predetermined lower threshold in order to suppress updating in environments where the signal is dominated by intrinsic microphone noise.
  • Another example might be the use of this capability to disable adaptation above a predetermined high threshold in order to suppress updating in an environment dominated by wind noise or in an environment where the signal might be distorted due to a level exceeding the dynamic range of the hearing aid.
  • the update/stall-circuit 72 outputs an enable-signal if the comparator 77 indicates that the signal envelope is equal to or below the threshold, and outputs a disable-signal if the comparator indicates that the signal envelope is above the threshold.
  • the update and stall criteria could as well be reversed, i.e. the update/stall-circuit 72 outputs a disable-signal if the comparator 77 indicates that the signal envelope is equal to or below the threshold, and outputs an enable-signal if the comparator indicates that the signal envelope is above the threshold.
  • detector 71 of DSD calculates a correlation coefficient of the input signals and the DSD concludes "desired signal” if the correlation coefficient reaches a certain value and the second control parameter 80 is adjusted accordingly.
  • Fig. 10 shows an embodiment of a desired signal detector 270 according to still another embodiment of the present invention.
  • the DSD 270 may be implemented in a signal processing apparatus 100, 200 as described with reference to Figures 1 and 2 .
  • the circuit structure of the DSD 270 comprises a correlation calculator 220 which calculates a correlation coefficient between the two input signals 20, 30 and submits this correlation coefficient to comparator 77.
  • the comparator 77 also receives a certain level signal from level generator 210, compares these two input signals and submits a comparison result to the update/stall-circuit 72 which produces the second control parameter 80.
  • the correlation calculator 220 it is determined whether the input signals (first and second microphone signal) 20, 30 are generated from the same sound source or not. For example, when the hearing aid is operated in a silent environment, each of the microphone signals contains only noise generated by the respective microphone itself. Thus, in this case, the input signals are generated by independent and thus non-correlated signal sources, namely the individual microphones. In this and other cases the correlation coefficient indicates whether at least one of the microphone signals is dominated by noise or distortion. For example, the adaptation may be stalled by transmitting a respective second control parameter 80 when at least one of the input signals 20, 30 is dominated by noise or distortion, so that comparator 77 detects falling of the correlation coefficient under the signal level generated by level generator 210.
  • Level generator 210 is at least similar to level generator 110, and the generated signal level that serves as a threshold in the comparator 77 may also be adjusted depending on the desired performance and choice of the hearing aid designer.
  • a correlation detector for detection of non-correlated first and second input signals and for generation of a control signal is provided by way of example.
  • the update and stall criteria may be adjusted based on the value of the correlation coefficient.
  • the update/stall-circuit 72 outputs an enable-signal if the comparator 77 indicates that the correlation coefficient is equal to or below the threshold, and outputs a disable-signal if the comparator indicates that the correlation coefficient is above the threshold.
  • the update and stall criteria could as well be reversed, i.e. the update/stall-circuit 72 outputs a disable-signal if the comparator 77 indicates that the correlation coefficient is equal to or below the threshold, and outputs an enable-signal if the comparator indicates that the correlation coefficient is above the threshold.
  • the selection between desired signals and undesired noise is implemented in various ways using different detectors 71 in the DSD 70 depending on the type of signal.
  • the selection may be based, e.g. on statistical analysis, frequency shaping, detection of certain non-linearities, or others.
  • Fig. 7 shows a signal diagram illustrating a signal envelope and an 11 % percentile estimator result of a single speaker over a time period of 20 seconds.
  • the signal envelope and the 11 % percentile estimator result have been achieved by using a digital implementation with a 32 kHz sampling frequency and a 24 Bit ADC and a desired signal detector 70 with an signal envelope circuit 74 and a percentile estimator 76 according to the present invention.
  • the improved behavior of parameter omni when applying a signal processing apparatus according to the present invention may simply be recognized for a skilled person.
  • the full line the parameter omni is adjusted between 1 and 0.97 over the entire time frame of 20 seconds.
  • a prior art directional controller without DSD adjusts the directional parameter omni in the same situation as shown by the dotted line in Fig. 8 .
  • the directional controller stays in omni-directional mode.
  • the frequency spectrum of the first and second microphone signals 20, 30 may be divided by band-split filters (not shown), respectively, into a number, e.g. three, of channels with respective limited frequency ranges.
  • band-split filters not shown
  • Each of the band-limited channels is then handled by a corresponding signal processing apparatus 100, 200, whereby each signal processing apparatus operates in a band-limited channel.
  • This system allows the directional characteristics to be different among these channels, such that the analysis by which signals are classified as desired and undesired signals and the directional characteristics is adjusted is done independently in respective frequency bands.
  • embodiments or features of embodiments described above may be combined in any combination useful in a directional system for minimizing noise.

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Claims (17)

  1. Signalverarbeitungsvorrichtung (100, 200) für ein Hörgerät mit einer steuerbaren Richtcharakteristik, die eine Richtungssteuereinheit (10), die dazu ausgelegt ist, ein erstes und ein zweites Mikrophonsignal Xfront, Xback (20, 30) zu empfangen, ein Ausgangssignal Y (40) auszugeben und das rückgekoppelte Ausgangssignal (60) als einen ersten Steuerparameter zu empfangen, um die Richtcharakteristik so einzustellen, dass das Ausgangssignal (40) minimiert wird, und einen Signalanalysator (70), der dazu ausgelegt ist, festzustellen, ob das erste und/oder das zweite Mikrophonsignal gewünschte Signale enthält, umfasst,
    dadurch gekennzeichnet, dass
    die Richtungssteuereinheit dazu ausgelegt ist, das Einstellen der Richtcharakteristik anzuhalten, falls der Signalanalysator gewünschte Signale in dem ersten und/oder dem zweiten Mikrophonsignal erfasst.
  2. Signalverarbeitungsvorrichtung nach Anspruch 1, wobei der Signalanalysator (70) einen Detektor (71), der dazu ausgelegt ist, das erste und/oder das zweite Mikrophonsignal (20, 30) zu empfangen und festzustellen, ob die empfangenen Mikrophonsignale gewünschte Signale enthalten, und eine Aktualisierungs-/Anhalteschaltung (72), die dazu ausgelegt ist, den Ausgang des Detektors (71) zu empfangen und einen zweiten Steuerparameter (80) auszugeben, der der Richtungssteuereinheit ermöglicht, die Richtcharakteristik einzustellen, falls unerwünschte Signale erfasst worden sind, umfasst.
  3. Signalverarbeitungsvorrichtung nach Anspruch 1 oder 2, wobei die Richtungssteuereinheit (10) dazu ausgelegt ist, die Richtcharakteristik durch Einstellen eines internen Steuerparameters (95) einer adaptiven Richtfunktion (50) einzustellen.
  4. Signalverarbeitungsvorrichtung nach Anspruch 3, wobei die adaptive Richtfunktion durch die folgende Formel definiert ist: Y = X front * 1 - omni * e - jωT + X back * omni - e - jωT
    Figure imgb0005
    wobei omni der interne Steuerparameter (95) ist und T eine vorgegebene akustische Verzögerung ist.
  5. Signalverarbeitungsvorrichtung nach einem der vorhergehenden Ansprüche, die ferner eine Parametersteuereinheit (90) umfasst, die dazu ausgelegt ist, ein rückgekoppeltes Ausgangssignal (60) und einen zweiten Steuerparameter (80) zu empfangen und den internen Steuerparameter (95) auszugeben, wobei die Parametersteuereinheit dazu ausgelegt ist, den internen Steuerparameter (95) einzustellen, um das rückgekoppelte Ausgangssignal (60) durch Anwenden eines Minimierungsalgorithmus zu minimieren, falls die Richtungssteuereinheit durch den zweiten Steuerparameter (80) freigegeben wird.
  6. Signalverarbeitungsvorrichtung nach einem der Ansprüche 2 bis 5, wobei der Detektor (71) des Signalanalysators (70) umfasst:
    - eine Signalhüllkurven-Schaltung (74), die dazu ausgelegt ist, das erste und/oder das zweite Mikrophonsignal (20, 30) zu empfangen und eine Signalhüllkurve der Mikrophonsignale zu erzeugen;
    - eine Perzentilschätzeinrichtung (76), die dazu ausgelegt ist, ein Perzentilschätzergebnis der Signalhüllkurven zu erzeugen; und
    - einen Komparator (77), der dazu ausgelegt ist, Signalpegel der Signalhüllkurve und des Perzentilschätzergebnisses zu vergleichen;
    wobei die Aktualisierungs-/Anhalteschaltung (72) dazu ausgelegt ist, als zweiten Steuerparameter ein Freigabesignal auszugeben, falls der Vergleich angibt, dass die Signalhüllkurve gleich oder niedriger als das Perzentilschätzergebnis ist, und als zweiten Steuerparameter ein Sperrsignal auszugeben, falls der Vergleich angibt, dass die Signalhüllkurve über dem Perzentilschätzergebnis liegt.
  7. Signalverarbeitungsvorrichtung nach einem der Ansprüche 2 bis 5, wobei der Detektor (71) des Signalanalysators (170) umfasst:
    - eine Signalhüllkurvenschaltung (74), die dazu ausgelegt ist, das erste und/oder das zweite Mikrophonsignal (20, 30) zu empfangen und eine Signalhüllkurve der Mikrophonsignale zu erzeugen;
    - einen Pegelgenerator (110), der dazu ausgelegt ist, einen Signalpegel als einen Schwellenwert zu erzeugen; und
    - einen Komparator (77), der dazu ausgelegt ist, festzustellen, ob die Signalhüllkurve über oder unter dem Schwellenwert liegt;
    wobei die Aktualisierungs-/Anhalteschaltung (72) dazu ausgelegt ist, in Abhängigkeit von dem Komparatorergebnis entweder ein Freigabe- oder ein Sperrsignal auszugeben.
  8. Signalverarbeitungsvorrichtung nach einem der Ansprüche 2 bis 5, wobei der Detektor (71) des Signalanalysators (70) dazu ausgelegt ist, zwischen Signalen auszuwählen, die erwünschte und unerwünschte Signale enthalten, indem eine statistische Analyse und eine Frequenzformung verwendet werden oder indem bestimmte Nichtlinearitäten in den Mikrophonsignalen erfasst werden.
  9. Signalverarbeitungsvorrichtung nach einem der Ansprüche 2 bis 5, wobei der Detektor (71) des Signalanalysators (70) eine Filterschaltungsanordnung enthält, die dazu ausgelegt ist, die Mikrophonsignale in bestimmte Frequenzbänder zu unterteilen, wobei die Frequenzbänder aus einem ersten Frequenzbereich, der einem Bereich erwünschter Signale entspricht, und aus einem zweiten Frequenzbereich, der einem Bereich unerwünschter Signale entspricht, bestehen, wobei die Aktualisierungs-/Anhalteschaltung (72) dazu ausgelegt ist, ein Freigabesignal auszugeben, falls die Erfassung nur Signale in dem zweiten Frequenzbereich angibt, und ein Sperrsignal auszugeben, falls die Erfassung Signale in dem ersten Frequenzbereich angibt.
  10. Signalverarbeitungsvorrichtung nach einem der Ansprüche 2 bis 5, wobei der Detektor (71) des Signalanalysators (70) dazu ausgelegt ist, einen Korrelationskoeffizienten der Mikrophonsignale zu berechnen, wobei die Aktualisierungs-/Anhalteschaltung (72) dazu ausgelegt ist, in Abhängigkeit vom Wert des Korrelationskoeffizienten entweder ein Freigabe- oder ein Sperrsignal auszugeben.
  11. Signalverarbeitungssystem, das eine Anzahl von Signalverarbeitungsvorrichtungen nach einem der vorhergehenden Ansprüche besitzt und ferner Bandunterteilungsfilter umfasst, die dazu ausgelegt sind, die ersten und zweiten Mikrophonsignale (20, 30) in eine Anzahl von bandbegrenzten Signalen zu unterteilen, wobei jede entsprechende Signalverarbeitungsvorrichtung dazu ausgelegt ist, eines der bandbegrenzten Signale zu empfangen und eine Richtcharakteristik durch getrenntes Verarbeiten des jeweiligen bandbegrenzten Signals zu verwenden.
  12. Verfahren zum Steuern der Richtcharakteristik eines Hörgeräts, das beabstandete erste und zweite Mikrophone und eine Richtungssteuereinheit, die die von dem ersten bzw. dem zweiten Mikrophon gelieferten ersten bzw. zweiten Mikrophonsignale empfängt und ein Ausgangssignal ausgibt, wobei das Ausgangssignal durch Kombinieren des ersten und des zweiten Mikrophonsignals in Übereinstimmung mit der Richtcharakteristik erzeugt wird, enthält, wobei das Verfahren ferner die folgenden Schritte umfasst:
    - Erfassen (510), ob das erste und/oder das zweite Mikrophonsignal gewünschte Signale enthalten;
    - Empfangen des rückgekoppelten Ausgangssignals als einen ersten Steuerparameter; und
    - Einstellen (530) der Richtcharakteristik, um das Ausgangssignal zu minimieren;
    gekennzeichnet durch:
    - Anhalten des Einstellens der Richtcharakteristik, falls in dem Erfassungsschritt erwünschte Signale erfasst werden.
  13. Verfahren nach Anspruch 12, wobei das Ausgangssignal durch Anwenden einer adaptiven Richtfunktion erzeugt wird, in der das erste und/oder das zweite Mikrophonsignal in Übereinstimmung mit einem internen Steuerparameter verzögert oder gedämpft werden und dann mit dem Ausgangssignal kombiniert werden und in der der interne Steuerparameter eingestellt wird, um das Ausgangsignal nur dann zu minimieren, wenn unerwünschte Signale erfasst worden sind.
  14. Verfahren nach Anspruch 12 oder 13, das ferner umfasst:
    - Erzeugen (610) einer Signalhüllkurve eines Eingangssignals, das dem ersten oder dem zweiten Mikrophonsignal oder einer Summe aus dem ersten und dem zweiten Mikrophonsignal entspricht;
    - Berechnen (620) eines Perzentilschätzergebnisses der Hüllkurve;
    - Vergleichen (630) von Signalpegeln der Signalhüllkurve und des Perzentilschätzergebnisses;
    - Aktualisieren (660) der Einstellung des internen Steuerparameters, falls die Vergleichsoperation ergibt, dass die Signalhüllkurve gleich oder niedriger als das Perzentilschätzergebnis ist;
    - Anhalten (650) der Einstellung des internen Steuerparameters, falls die Vergleichsoperation ergibt, dass die Signalhüllkurve über dem Perzentilschätzergebnis liegt.
  15. Hörgerät, das das Verfahren nach einem der Ansprüche 12 bis 14 ausführt.
  16. Hörgerät (220) mit einer steuerbaren Richtcharakteristik, das eine Signalverarbeitungsvorrichtung oder ein Signalverarbeitungssystem nach einem der Ansprüche 1 bis 11, beabstandete erste und zweite Eingangswandler (Fmic, Bmic), die ein erstes bzw. ein zweites Mikrophonsignal (20, 30) liefern, und einen Ausgangswandler (210) zum Aussenden von Schallsignalen in Reaktion auf das Ausgangssignal umfasst.
  17. Computerprogrammprodukt, das Computerprogrammcode enthält, der dann, wenn er auf einem Computer oder einem digitalen Signalverarbeitungssystem ausgeführt wird, dem Computer oder dem digitale Signalverarbeitungssystem ermöglicht, ein Verfahren nach einem der Ansprüche 12 bis 14 auszuführen.
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